Hi Lloyd, as you are using Aastra phone 9143i behind NAT, plesse enable "rport" under "Network" settings of Aastra phone,and restart the phone, most likely you should be able to bet back to work unless your SMC Gateway really sucks.<br>
Please advise if this helps.<br><br>Chris<br><br><br><div class="gmail_quote">On Fri, Apr 9, 2010 at 3:40 PM, Aloysius Lloyd <span dir="ltr"><<a href="mailto:lloyd.aloysius@gmail.com">lloyd.aloysius@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I could not find a SIP ALG Setting. I setup the <span style="font-family: arial,sans-serif; font-size: 13px; border-collapse: collapse;">sip-force-expires and ping for the user directory. Only one time registering then lost the connection.</span><div>
<font face="arial, sans-serif"><span style="border-collapse: collapse;"><br></span></font></div><div><font face="arial, sans-serif"><span style="border-collapse: collapse;">Here is the sofia profile internal status</span></font></div>
<div><font face="arial, sans-serif"><span style="border-collapse: collapse;"><br></span></font></div><div><font face="arial, sans-serif"><span style="border-collapse: collapse;"><div>
Call-ID: 7307ef8fa6044407</div><div>User: <a href="mailto:202@abc.com" target="_blank">202@abc.com</a></div><div>Contact: "Mike Derouin" <sip:202@A.B.C.D:5060;transport=udp;srcadr=<a href="http://192.168.1.51:5060" target="_blank">192.168.1.51:5060</a>></div>
<div>Agent: Aastra 9143i/<a href="http://2.5.2.30" target="_blank">2.5.2.30</a></div><div>Status: Registered(UDP)(unknown) EXP(2010-04-09 15:31:21)</div><div>Host: TestSrv</div><div>IP: A.B.C.D</div>
<div>Port: 5060</div><div>Auth-User: 202</div><div>Auth-Realm: <a href="http://abc.com" target="_blank">abc.com</a></div><div>MWI-Account: <a href="mailto:202@abc.com" target="_blank">202@abc.com</a></div>
<div><br></div><div>Please let me know how to fix this issue.</div>
<div><br></div><div>Thanks</div><div>Lloyd</div></span></font><div><div></div><div class="h5">
<br><br><div class="gmail_quote">On Fri, Apr 9, 2010 at 1:37 AM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="word-wrap: break-word;">I dont think there is a lowest value, but 30 seconds is reasonable in most cases.<div>You can also add a ping parameter with value 30, in the user config with the variables.</div><div>
The result is a SIP OPTIONS sent to the phone every X sec.</div>
<div>That's quite the equivalent of qualify=yes in Asterisk.</div><div><div><br><div> <span style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div style="word-wrap: break-word;">
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<div><font face="'Helvetica Neue'"><font color="#1c00ff">David Ponzone </font><font color="#000000" size="3"><span style="font-size: 12px;">Direction Technique</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></font></font></div>
<div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">tel: 01 74 03 18 97</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">gsm: 06 66 98 76 34</span></font></font></div>
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<div><font face="'Helvetica Neue'" size="3"><span style="font-size: 13px;"><div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><a>www.ipeva.fr</a></span><span style="color: rgb(101, 104, 149);"> - <span style="color: rgb(0, 34, 243); text-decoration: underline;"><a>www.ipeva-studio.com</a></span></span></div>
<div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><br></span></div><div style="margin: 0px; color: rgb(0, 34, 243);">
<span><div style="margin: 0px; text-align: justify; color: rgb(192, 192, 192);"><i>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. </i><b><i>IPeva</i></b><i> décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.</i></div>
<div style="text-decoration: underline; text-align: justify;"><font color="#c0c0c0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br></div></span><br> </div><br></div><div><div>
Le 09/04/2010 à 03:31, Aloysius Lloyd a écrit :</div>
<div><div></div><div><br><blockquote type="cite">what is the lowest value I can use for sip-force-expires ?<div><br></div><div>Thanks</div><div>Lloyd<br> <br><br><div class="gmail_quote">On Thu, Apr 8, 2010 at 9:29 PM, Aloysius Lloyd <span dir="ltr"><<a href="mailto:lloyd.aloysius@gmail.com" target="_blank">lloyd.aloysius@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> <div><br></div><div>Yes Asterisk config using the qualify=yes. </div><div><br></div><div>I will modify the Registration time out by 50sec and see how it is behaving.</div>
<div><br></div><div>Thanks</div><div>Lloyd</div><div> <div></div><div><div> <br></div><div><br> <br><br><div class="gmail_quote">On Thu, Apr 8, 2010 at 7:23 PM, David Ponzone <span dir="ltr"><<a href="mailto:david.ponzone@gmail.com" target="_blank">david.ponzone@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> <div style="word-wrap: break-word;">On most low-end routers, the NAT table will expire UDP translations after 60 sec.<div>
Did you configure your phones to send a NAT keep-alive every X seconds, with X < 60 ?</div><div>You can also use sip-force-expires on the FS side.</div> <div><br></div><div>In your Asterisk config, do you use qualify=yes ?</div>
<div><br></div><div><div><span style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;"><div style="word-wrap: break-word;">
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<div><font face="'Helvetica Neue'"><font color="#1c00ff">David Ponzone </font><font color="#000000" size="3"><span style="font-size: 12px;">Direction Technique</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">email: <a href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></font></font></div>
<div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">tel: 01 74 03 18 97</span></font></font></div><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">gsm: 06 66 98 76 34</span></font></font></div>
<div><font face="'Helvetica Neue'"><br></font></div><div><font color="#1c00ff" face="'Helvetica Neue'">Service Client<span> </span></font><font face="'Helvetica Neue'"><font color="#ff0000">IP</font></font><font color="#1c00ff" face="'Helvetica Neue'">eva</font></div>
<div><font color="#1c00ff" face="'Helvetica Neue'"><span style="color: rgb(0, 0, 0); font-family: Helvetica;"><div><font face="'Helvetica Neue'"><font size="3"><span style="font-size: 13px;">tel: 0811 46 26 26</span></font></font></div>
<div><font face="'Helvetica Neue'" size="3"><span style="font-size: 13px;"><div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><a>www.ipeva.fr</a></span><span style="color: rgb(101, 104, 149);"> - <span style="color: rgb(0, 34, 243); text-decoration: underline;"><a>www.ipeva-studio.com</a></span></span></div>
<div style="margin: 0px; color: rgb(0, 34, 243);"><span style="text-decoration: underline;"><br></span></div><div style="margin: 0px; color: rgb(0, 34, 243);">
<span><div style="margin: 0px; text-align: justify; color: rgb(192, 192, 192);"><i>Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. </i><b><i>IPeva</i></b><i> décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.</i></div>
<div style="text-decoration: underline; text-align: justify;"><font color="#c0c0c0"><i><br></i></font></div></span></div></span></font></div></span></font></div></div></span><br></div></span><br> </div><br><div><div>Le 08/04/2010 à 23:00, Aloysius Lloyd a écrit :</div>
<br><blockquote type="cite"><div><div></div><div>Hi All,<div><br></div><div>I am having lots of problem with NAT and FreeSWITCH.</div><div><br></div><div>FreeeSWITCH running Public IP.</div><div><br></div><div> Phones Connecting to FreeeSWITCH through NAT. Cable Provider Router/Modem : SMC Gateway.</div>
<div><br></div><div>The Phone Losing the Registrations. The Environment working for Asterisk without any problem.</div><div> <br> </div><div>Please note I am not comparing Asterisk Vs FreeSWITCH. I am trying make FreeSWITCH work.</div>
<div><br></div><div>How to solve this problem.</div><div><br clear="all">Thanks,<br>Lloyd </div></div></div><div> _______________________________________________<br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
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